1. Field of the Invention
The present invention relates to an impulse response collecting method, a sound effect adding apparatus, and a record medium that allow reverberation to be added corresponding to a real apparatus or a real space.
2. Description of the Related Art
As an apparatus that adds a sound effect to an audio signal, a reverberator is known. The reverberator is used to add reverberation to an audio signal in for example a recording studio so that listeners can have a spatial impression and a deep impression. When reverberation is added to an audio signal that has been recorded in a studio or the like, a sound effect performed in a hall and a special effect can be added to the audio signal.
Formerly, to add reverberation to an audio signal, sound was recorded in for example a hall where reverberation was obtained. Alternatively, such as a steel-plate echo apparatus was used to obtain a reverberative effect. In a recent reverberator, such an effect is electrically accomplished. More recently, as digital technologies have advanced, an apparatus that digitally synthesizes reverberation is becoming common.
When reverberation is added to an audio signal by a digital process, for example a recursive digital filter is used. With the recursive digital filter, an input digital audio signal is attenuated and recurred. Thus, reverberation is generated. The generated reverberation is mixed with the original digital audio signal. In reality, initial reflection sound is added at a position delayed by a predetermined time period against direct sound. After a predetermined time period, reverberation is added. The delay time period of the reverberation against the direct sound is referred to as pre-delay. By adjusting the reverberation time, adding sub-reverberation, and finely adjusting the level of reverberation, a variety of types of sound can be generated.
Reverberation in a real hall has a complicated waveform because of various reflections and interferences of sound due to the shape of the hall and the position of a sound source. However, as described above, in the method of which an original digital audio signal is processed with a filter, since the original signal is simply attenuated, the listeners of the resultant signal have an artificial impression about the generated sound. On the other hand, in the method of which an original digital audio signal is recurred by a filter process, after an input signal ceases, since the final pitch of reverberation is equal to the pitch of the inner feed-back loop of a recursive filter. Thus, in this method, natural and high quality reverberation cannot be obtained.
As a method for reproducing a rear sound field along with a front sound field using left and right sound sources is becoming common. This reproducing method is referred to as surround system. In the surround system, a sound field of a movie theater is reproduced. For example, in front of the listener, left and right channels (F-L and F-R) are disposed. Behind the listener, left and right channels (R-L and R-R) are disposed. In addition to the four channels, at the center of the front of the listener, another channel (referred to as C channel) is disposed. The C channel is used to reproduce speeches of actors and actresses. As an optional channel, one channel is disposed at any position. The optional channel is used to reproduce a sound in the ultra low band. Since the information amount of the ultra low band is 1/10 that of each of the other channels, the channel structure of 5 channels plus ultra low band is referred to as 5.1 channel structure.
In addition to movies, when a conventional music performance that has been recorded in the surround system is reproduced in a relevant system, the listener can enjoy the performance with more presence. In the case of a music performance, four channels F-L, F-R, R-L, and R-R are normally used. However, the C channel is not used.
When sound is recorded in a real hall or the like, reverberation can be obtained more naturally. However, in a real hall, parameters with respect to reverberation (such as reverberative time) cannot be varied. In addition, the positions and types (characteristics) of microphones cannot be quickly changed. Moreover, many apparatuses are required. In addition, due to noise of air conditioners, S/N ratio of sound is low. Therefore, there are many problems to be solved in the related art.
Likewise, a mechanical reverberator such as a steel-plate echo apparatus or a spring echo apparatus may be used. However, such apparatuses have problems of aged tolerance and necessity of maintenance. These problems become critical for an apparatus that cannot be obtained due to out-of-fabrication. In addition, such apparatuses are adversely affected by vibration and external noise. The reverberation time cannot be freely adjusted. Moreover, such apparatuses do not have good reproducibility. Furthermore, the weight and size of these apparatuses are large and S/N ratio of obtained sound is not high.
On the other hand, a method for generating reverberation in a real hall or with a steel-plate echo apparatus, collecting an impulse response corresponding to the generated reverberation, and performing a convolution calculation for the collected impulse response and the input data by a filter process has been proposed. Thus, more natural reverberation corresponding to an impulse response of a real space or an apparatus can be obtained.
FIG. 1 shows an example of a structure for performing a convolution calculation for an impulse response in time axis direction using an FIR (Finite Impulse Response) filter. Coefficients of an impulse response are required corresponding to samples of an input digital audio signal. Thus, when the impulse response data of 219 points (524,288 points≈512 k points) is obtained with a digital audio signal at a sampling frequency of 48 kHz, the reverberation time becomes around 10 seconds.
In FIG. 1, a digital audio signal is supplied from a termina 310. The number of quantizing bits of the digital audio signal is for example 24. The sampling frequency of the digital audio signal is 48 kHz. The input signal is supplied to 512 k delaying circuits 311 connected in series. Each of the 512 k delaying circuits 311 has a delay of one sample. Output signals of the individual delaying circuits 311 are supplied to respective coefficient multiplying devices 312. Impulse response data of the first point to 512 k-th point is supplied to the delaying circuits 311 with 24 quantizing bits. The coefficient multiplying devices 312 multiply respective output signals of the delaying circuits 311 by respective impulse response data. The multiplied results are added by an adding device 313. The added result is supplied as reverberation data against the input data to a terminal 314.
In the method for performing convolution calculations for impulse response data in time axis direction, a huge number of delaying circuits 311 and coefficient multiplying devices 312 are required.
To solve such a problem, as shown in FIG. 2, a method for converting an input audio digital audio signal and impulse response data into frequency element data corresponding to Fourrier transform method has been proposed.
Referring to FIG. 2, an input digital audio signal is supplied from a terminal 320. Data for samples corresponding to a required reverberation time (namely, data for 512 k points) is stored in a memory 321. Data stored in the memory 321 is supplied to an FFT (Fast Fourrier Transform) circuit 322. The FFT circuit 322 performs fast Fourrier transform for the data received from the memory 321 and outputs frequency element data of for example 0.1 Hz. Likewise, impulse response data is supplied from a terminal 323. The impulse response data is stored in a memory 324. The impulse response data is supplied to an FFT circuit 325. The FFT circuit 355 performs fast Fourier transform for the impulse response data received from the memory 324 and outputs frequency element data. Since the impulse response data is known, the FFT 325 and the memory 324 may be composed of a ROM 326.
Output data of the FFT circuits 322 and 325 is supplied to a multiplying device 327. The multiplying device 327 multiplies the output data of the FFT circuit 322 by the output data of the FFT circuit 325 in such a manner that the frequency components thereof match. The multiplied result is supplied to an IFFT circuit 328. The IFFT circuit 328 performs inversely fast Fourrier transform for the data received from the multiplying device 327 and outputs the resultant data as time axis data to a terminal 329.
In this method, the hardware scale is smaller than that of the convolution calculation method on time axis. However, since input data corresponding to the required reverberation time should be temporarily stored to the memory 321, a delay of output data against input data becomes large.
To solve the above-described problems about the convolution calculation process for an impulse response, a method for dividing impulse response data on time axis and performing a convolution calculation process for input data corresponding to the divided blocks of the impulse response data has been proposed (as Japanese Patent Publicized Publication No. 8-501667). However, in this method, it is not easy to accurately collect impulse response. This related art reference does not mention how to collect an impulse response.
In other words, the reverberation time has been defined as a time period after sound ceases until the sound pressure level attenuates by 60 dB. The reverberation should be recorded in all the level range. Since the reverberation should be generated with signals including a very low level signal, noise tends to enter the reverberation. In addition, it is very difficult for the user to record reverberation in a real hall.
When a music performance is recorded in the surround system, a reverberation corresponding to a sound field formed with sounds of the four channels should be added. Conventionally, two reverberation adding apparatuses that output monaural/stereo signals are used. In addition, to obtain a high quality reverberation, digital reverberation adding apparatuses have been used.
For example, when one sound source on a stage is recorded, the first reverberation adding apparatus adds a reverberation corresponding to F-L and F-R channels that are on the stage side against the listener. In addition, the second reverberation adding apparatus adds a reverberation corresponding to the R-L and R-R channels that are on the rear side against the listener.
Since a reverberation varies position by position, the two reverberation adding apparatuses should be independently set. Thus, to record a sound source in the surround system, since two reverberation adding apparatuses should be used, their operations are inconveniently complicated.
The reverberation as such as in the hall, the sound is reflected and interfered with the structure of the hall or the position of sound sources then, the waveform becomes more complicated. As mentioned before, attenuated waveforms can be only gained by a filter processing method of the former digital audio signal so, there is a problem of hardly eliminating an artificial impression with the sound
On the other hand, a sound source having a particular sound field should be recorded in stereo. In this case, reverberation adding apparatuses corresponding to the stereo input signals are used. However, since conventional reverberation adding apparatuses corresponding to stereo input signals artificially generate stereo sounds, their sounds are unnatural.
To obtain a more natural reverberation, a sound source may be recorded in a real hall. However, in this case, apparatuses should be set and operated in the hall. In addition, the hall may not be available on the desired date and time. Moreover, in this case, know-how is required for setting microphones. In addition, to lower dirk noise, an air-conditioner and so forth should be stopped.